Closed khdevel closed 8 years ago
Hrmmm, thanks for the issue submission + taking a look. I just added a Travis build for the project a week ago and have tests where a SIP call is made between two boxes so you might want to take a look at the .travis.yml file and see how it's run, in the meanwhile.
I'll take another look this coming week, I'm on the road this weekend, so I might not be able to give it a lot of attention until then.
I'll update soon!
On Sat, Feb 27, 2016, 12:39 PM kamil notifications@github.com wrote:
Hi,
I like your Dockerfile for the asterisk. I run it and it seems to work... expect that it does not have a sip port open. I saw that there is a miss of the _chansip in the configuration. I checked also the make menuconfig and it is not set.
Of course I tried to set it by myself but without the success. The command: _menuselect/menuselect --enable chansip menuselect.makeopts did not work for me, so I made a changes by some sed like:
RUN sed -i -e 's/\ chan_sip//' menuselect.makeopts RUN sed -i -e 's/G711_NEW_ALGORITHM/res_crypto res_http_websocket G711_NEW_ALGORITHM/' menuselect.makeopts RUN sed -i -e 's/res_fax_spandsp\ res_http_websocket/res_fax_spandsp/' menuselect.makeopts
but it still does not work - the port 5060 is not open like it is with IAX in your default config.
Could you please help me and update the Dockerfile for the SIP protocol too.
Thank you!
— Reply to this email directly or view it on GitHub https://github.com/dougbtv/docker-asterisk/issues/40.
I'm still mobile, but check out the latest build: https://travis-ci.org/dougbtv/docker-asterisk
core show modules
in the build shows chan_sip -- what's your docker run
command look like and what does your core show modules look like?
On Sat, Feb 27, 2016, 5:09 PM Doug Smith douglaskippsmith@gmail.com wrote:
Hrmmm, thanks for the issue submission + taking a look. I just added a Travis build for the project a week ago and have tests where a SIP call is made between two boxes so you might want to take a look at the .travis.yml file and see how it's run, in the meanwhile.
I'll take another look this coming week, I'm on the road this weekend, so I might not be able to give it a lot of attention until then.
I'll update soon!
On Sat, Feb 27, 2016, 12:39 PM kamil notifications@github.com wrote:
Hi,
I like your Dockerfile for the asterisk. I run it and it seems to work... expect that it does not have a sip port open. I saw that there is a miss of the _chansip in the configuration. I checked also the make menuconfig and it is not set.
Of course I tried to set it by myself but without the success. The command: _menuselect/menuselect --enable chansip menuselect.makeopts did not work for me, so I made a changes by some sed like:
RUN sed -i -e 's/\ chan_sip//' menuselect.makeopts RUN sed -i -e 's/G711_NEW_ALGORITHM/res_crypto res_http_websocket G711_NEW_ALGORITHM/' menuselect.makeopts RUN sed -i -e 's/res_fax_spandsp\ res_http_websocket/res_fax_spandsp/' menuselect.makeopts
but it still does not work - the port 5060 is not open like it is with IAX in your default config.
Could you please help me and update the Dockerfile for the SIP protocol too.
Thank you!
— Reply to this email directly or view it on GitHub https://github.com/dougbtv/docker-asterisk/issues/40.
@khdevel -- glad to see that worked! definitely submit a PR, happy to take a look with the changes, should be a-OK as it is. Thanks for digging through it!
I'm going to close this out for now. If you've got another recommendation, I'm happy to reopen it and look further however I currently believe chan_sip to be enabled.
Hi, Sorry for no reply from my side, something else disturbed my mind and I did not follow this topic... I will test it this week I hope and I believe that it works as you wrote above :)
Thank you very much!
Thanks @khdevel -- I'll open the issue back up and happy to collab if you're not getting the results you expect for sure.
@khdevel -- I was working on the asterisk13 image and I noticed that it didn't have chan_sip! Just pjsip!
So, I fixed that merged in with #49 . Now...... now you should have some luck! If I don't hear from you for a couple days, I'll close 'er out, I think I finally found what you had tried to show me, my bad for not picking up on it sooner.
Hi,
I have tested you fix and it seems to work fine. I did not use your code from GitHub but I used the image from the Docker's repository dougbtv/asterisk13. My host was Centos7 and after I run the container I could register my two SIP phones and make a call between them. No modifications were needed except those I had to make to register and make a call (sip.conf and extensions.conf).
Thank you very much!
Thanks Kamil for testing it! The pulled image is perfect, it should always be at HEAD of master, so it includes these changes. Appreciate you opening a ticket and working with me on it, for sure.
-Doug
No problem and sorry for delays :)
Have a nice coding!
Hi,
I like your Dockerfile for the asterisk. I run it and it seems to work... expect that it does not have a sip port open. I saw that there is a miss of the _chansip in the configuration. I checked also the make menuconfig and it is not set.
Of course I tried to set it by myself but without the success. The command:
did not work for me, so I made a changes by some sed like:
but it still does not work - the port 5060 is not open like it is with IAX in your default config.
Could you please help me and update the Dockerfile for the SIP protocol too.
EDITED
I run my build again and this:
worked for me! The sip works fine and the port is open. Do you think it is a good moment for some pull request? Or maybe you know how to change above sed on some better way?
Thank you!